search for: inaudible

Hello, I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission...

...amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the original signal present in a live recording merely degrades the playback system's ability to reproduce the audible signal. Because a perceptual coder removes inaudible signal content, the playback system's ability to convey audible music logically should improve. In short, a perceptual coder more properly codes an audio signal for passage through an audio system." Is this bullshit or an interesting thought? --- >8 ---- List archives: http://www....

Hi all and thanks for reading. I am experiencing a frustrating issue with asterisk where on some calls the volume suddenly drops to inaudible o completely fades away for a time. If you hold on long enough (20 to 30 seconds) the sound will come back. My asterisk server is on a public IP, and basically acts as a VoIP bridge receiving calls from my customers (all of whom use Grandstream GXW400X gateways on public IP's, no NAT) and send...

The results are at www.rational.co.za/speex.csv Each of the 11 quality settings is tested 3 times (narrow, wide and ultra wide band). Strangely narrow band quality 11 outperforms all wide band tests, but it can be due to my subsampling or some other inaudible effect like delaying. You can import it into Excel and sort it by SNR or other value. Divide the bits by 24 to get the bps. The patch is at www.rational.co.za/speexBatchTest.patch The complete source is at www.rational.co.za/speex-1.1.10a.tar.gz on a slow server. ----- Original Message -...

...ing rate is 48000. The source of the data is a microphone using ALSA. That platform is Ubuntu. The version of speex is 1.2rc1. When I run with the default settings, the audio seems OK. It's not clear if preprocess is actually doing anything. If I turn AGC on, the sound level becomes almost inaudible. When I print the control parameters (the defaults), I get: DENOISE = 1 AGC = 0 AGC_LEVEL = 1174011904 AGC_INCREMENT = 12 AGC_DECREMENT = -40 AGC_MAX_GAIN = 30 VAD = 0 DEREVERB = 0 NOISE_SUPPRESS = -15 QUESTION number 1: Is there a guide to "tuning" the AGC? Is there a document that te...

...cloud account) encoded with libopus 1.2 RC1. I used Windows binaries from free-codecs.com. I noticed that in the case of my selected music file (which is generally harsh on lossy codecs as it's necessary to use quite a high bitrate to make differences between the original and processed file inaudible), there is a long drop of stereo to mono (or something which sounds similar) at the beginning of the track and a few others much shorter during the whole track lenght. The selected bitrate is 32 kb/s stereo and everything else on default values. This is not happenig with libopus 1.1.x. Also whil...

Jean-Mark sarkasm. Jean-Markasm. (Bonus points for providing an actual noisy WAV! ^_^) On 30/10/2017 20:28, Jean-Marc Valin wrote: Hi, Before I comment on the graphics you posted to visualize the difference between two audio signals, I'd like to ask for your help in evaluating my JPEG encoder. I've encoded an image with JPEG and then computed the difference with the original. I then

...ing IIR?s or an FIR with some delay (sufficient delay that the truncation of impulse response does not matter much). With higher sampling rates, it gives us flexibility to design this filter such that the stopband attenuation is large enough (although not ?infinite dB) and the noise is filtered to inaudible levels. From: opus-bounces at xiph.org [mailto:opus-bounces at xiph.org] On Behalf Of Edwin van den Oetelaar Sent: Saturday, June 07, 2014 5:22 PM To: Andrew Lentvorski Cc: opus at xiph.org; Jean-Marc Valin Subject: Re: [opus] High Sampling Rates On Sat, Jun 7, 2014 at 10:58 AM, Andrew Lentvorski...

Hi. I've been browsing the archive on this topic and only found a few notes, all dating from a year ago (almost too precisely :) ) -- hope I haven't skipped the mails on that matter, sorry if I did. <p>Basically, I will get in the next few months a MOTU 896, that have 8 i/os in 24/96. I do pro sound recording, so it's more or less my business to have such a piece. Of

...I wanted to ask the list if this is expected/known behaviour. I wonder if it is possible to engineer a scenario (using existing apis) that would give one the benefits of _AUDIO mode (low delay, great fidelity, avoid voice-EQ) without this particular sensitivity that gives high bitrate for nearly inaudible noise floor. It's a lot to pay for near subliminal comfort noise. I'm prepared to hear that the answer is: choose VOIP mode ;) Thank you! Warm regards, KevinC

...and the download service is free) there is 7zip archive with 5 music files. The tune is "Nick Warren - Devil's Elbow" and is freely available from the author's Soundcloud account. I use this song for encoder testing because it requires quite a high bitrate for artifacts to be inaudible (especially the "gas leakage" kind of sound at the very beginning (repeated several times during the full song lenght) of the track. The content of the 7z file is the original track in WAV, 2 files encoded with libopus 1.1.5 at 32 and 48 kb/s (only with --bitrate 32 or --bitrate 48 op...

I've just installed RC2 and I'm very excited about the quality. It's so much better than MP3. This is the first version I've used since I just found out about Ogg Vorbis. I did notice that very high frequencies seem to be missing but since not many people can hear much above 18 KHz it's not much of an issue. I suppose this resolves the hiss problem so prevalent in MP3.

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

...ly drop: Instead I just get a "disconnect" (or number unobtainable) tone. Could this be the problem (i.e. there's no actual voltage drop happening to signal the call has ended)? Or is there some sort of other change in the line that I wouldn't detect audibly? Could it be that any inaudible voltage drop might be happening too quickly for zaptel to detect? What might I change in the source code to see if this is the case? Does nobody else in the UK use these cards? I'm sure that's not the case. So if you do use them, please stand up and be counted -- did you have to make any...

That sounds like the ideal app for me too. Fring requires we register with Fring and give them user id/password pair. In our case it did not work until we put a public IP on our Asterisk. I just bought WeePhone and I'll give it a try on the iPhone. Cheers, Enrique -----Mensaje original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] En

Monkey's Audio lossless compressor (currently win32 only, free but not open-source except decoder) author is thinking to implement a kind of audiophile-quality lossy compression which would filter "noise bits" that are hard to encode lossless but which are (or should be) inaudible and thus improve lossless compression (avg. 300-450kbps). I think that implementing something like this in future OggSquish could be very interesting for people who want very high-quality but are unsatisfied with lossless compression ratios. Here's the message from Monkey's Audio forum: __...

SUMMARY I'm impressed. RC3 withstood anything I could throw at it. "-q 3" is really Vorbis' sweet spot, almost perfect, and "-q 0" is eminently usable, with only marginal defects for normal usage. Already sent money, will do that again. * * * I spent at least 40 hours in the last week testing RC3. I selected a few fragments from pop, classic and jazz CDs in my

...rvice is free) there > is 7zip archive with 5 music files. The tune is "Nick Warren - Devil's > Elbow" and is freely available from the author's Soundcloud account. I > use this song for encoder testing because it requires quite a high > bitrate for artifacts to be inaudible (especially the "gas leakage" kind > of sound at the very beginning (repeated several times during the full > song lenght) of the track. The content of the 7z file is the original > track in WAV, 2 files encoded with libopus 1.1.5 at 32 and 48 kb/s (only > with --bitrate 3...

...The results are at www.rational.co.za/speex.csv >> >> Each of the 11 quality settings is tested 3 times (narrow, wide and >> ultra >> wide band). Strangely narrow band quality 11 outperforms all wide band >> tests, but it can be due to my subsampling or some other inaudible >> effect >> like delaying. >> >> You can import it into Excel and sort it by SNR or other value. Divide >> the >> bits by 24 to get the bps. >> >> The patch is at www.rational.co.za/speexBatchTest.patch >> The complete source is at www.ra...

Hello, Merry Christmas and happy new year for everyone. I've started porting Speex 1.2rc1 on a 32-bit microcontroller, so far everything goes fine for Norrowband modes. But when I'm trying to encode in wideband mode with quality higher than 5 then decode my file back, I get inaudible "thing" (just strong noise). - All frames are decoded with correct result (decoder function returns 0 for each decoded frame), so I guess encoded frames are correct. - Starting from quality 5 a low noise becomes audible and gets higher when quality is increased till reachin...